1. Technical Field
The present invention relates generally to a voice over Internet protocol (VoIP) gateway system and, in particular, to a VoIP gateway for supporting a VoIP call of a legacy private branch exchange (PBX) or a key phone system that has only a foreign exchange office (FXO), and to a method for controlling the same.
2. Related Art
A voice over Internet protocol (VoIP) gateway system transmits voice and facsimile data received from a public switched telephone network (PSTN) to an Internet protocol (IP) network after real time compression and protocol conversion. Also, a VoIP board built into the VoIP gateway system converts audio data used by the PSTN into data used in the IP network. That is, the VoIP board with a VoIP gateway function of enabling a telephone call over the Internet supports the H.323 V3 protocol.
A foreign exchange office (FXO), typically implemented in the form of a board, provides an office line interface for an exchange. A foreign exchange station (FXS), also implemented in the form of a board, provides an extension line interface. A VoIP processor, also implemented in the form of a board, converts the audio data received from the PSTN into data used in the IP network. A controller in the form of a media gateway control board (MGCB) controls the FXO, the FXS, and the VoIP processor, thereby totally controlling the entire operation of the VoIP gateway (VG).
The VoIP gateway, developed to form the VoIP network, is interlocked with a digital interface (E1/T1) or an analog interface (FXO, FXS, or E&M) of a legacy private branch exchange (PBX) or a key phone system. The E1/T1, a board mounted on the controller, is used when the VoIP gateway is connected to an extension line E1 or T1. The E1 and the T1 are well-known high-speed digital lines supporting transmission rates of 2.048 Mbps and 1.544 Mbps, respectively. While the E1/T1 or E&M provides a bi-directional service for supporting both transmission and reception functions, the FXS and the FXO supporting a unidirectional service have limitations in providing services. That is, when it is assumed that each of the FXS and the FXO can provide an eight-port service because the number of incoming calls is eight and the number of outgoing calls is eight, a total of 16 channels are supported. However, the number of channels that can serve the outgoing calls is restricted to eight. That is, it is not possible to use all of the 16 channels for the outgoing calls. Also, when the legacy PBX or the key phone system that has only an FXO is used according to the state of a site, the legacy PBX or the key phone system must be expanded or exchanged in order to manage the VoIP network using the above interfaces.
A VoIP network includes a common legacy PBX or key phone system that has an FXO and an FXS. The E1/T1 digital interface (or trunk) supporting both the transmission and reception functions or the E&M analog interface is used to establish the VoIP network interlocked with the legacy PBX or the key phone system using the VoIP gateway. In the case of an interface using an FXS and an FXO, the FXS supports a reception function, while the FXO supports a transmission function. Therefore, the legacy PBX or the key phone system that has both the FXS and the FXO is connected to the FXO and the FXS of the VoIP gateway to perform a VoIP service.
However, when the legacy PBX or the key phone system has only the FXO, it cannot support both transmission and reception functions. This is because the FXO and the FXS are unidirectional. For example, when a subscriber in a first region tries to transmit a call to the FXS of the VoIP gateway using the FXO of the key phone system in the VoIP network including the FXO/FXS analog interfaces, a call setup signal is converted into a VoIP packet and is transmitted to a VoIP gateway in a second region. The call transmitted to the FXO of the VoIP gateway terminates at the FXS of the legacy PBX. Therefore, if the PBX or the key phone system in the second region does not have an FXO, the transmitted call cannot terminate.
The legacy PBX and the key phone system supporting a direct inward dial (DID) numbering function can operate as if the incoming call is terminated at the FXO. However, because a VoIP gateway transfers a VoIP incoming call to the FXO, a function of transferring the VoIP incoming call to the FXS is required in order to interface with the legacy PBX and the key phone system without the FXO. Also, in order to transmit a call to an appropriate FXS of the VoIP gateway, digits received from a caller party must be translated so as to be suitable for a called party.